Re-invite Without SDP
Nitin Kapoor <[email protected]> Fri, 27 Nov 2009 16:06:15 -0500
| Newsgroups | gmane.ietf.sip |
|---|---|
| Message-ID | <[email protected]> |
Hello All, I am facing the issue with one of my customer at the time of unhold the call. I have configured an endpoint that is dropping calls when you place the call on hold. The network configuration is as follows: Cisco CallManager using SIP signaling > Cisco Unified Border Element (CUBE) using SIP signaling > MSX communicates SIP to CUBE and h.323 to HT_5850_Egress > PSTN When a phone on the Cisco CallManager places a call to a user on the PSTN the call goes through successfully. When the phone on the Cisco CallManager places that call on hold, the call gets dropped. The message I see in the SIP trace is '488 Not acceptable here'. If we do this same test from the Call Manager to the CUBE to the MSX but instead to L3_Egress so the signalling between the CUBE/MSX/PSTN is SIP the hold option works. *First IWF.. Where this option is not working. *Calling number 763.432.8682 Called number 763.577.3948 On this First Source UA has send the initial invite and which SBC has sent to termination END as h.323 and call is connected successully after the the mesages... Now when my Source UA has sent the invite to put that call on hold it sends "*re-invite with SDP with codec g.711u and media attribute = inactive*", and call goes on hold successfully, but now whenever they are trying to unhold the call and sending the another "*invite without sdp*" than my SBC/Nextone is sending "*488 unacceptable*" to my source UA. *And now the other option SIP(both legs)* Calling number 763.432.8682 Called number 1612.964.8862 Now in this scenario i have both the legs on sip and when my source UA is sending the "invite without SDP" to unhold this call then my my termination end is sending 200 OK with SDP to ACK that call and unhold scenario is working fine. I am also attaching the call flow for both the scenario. Please help me to find out the root cause of this issue. Thanks, Nitin Kapoor General Telecom _______________________________________________ Sip mailing list https://www.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use [email protected] for questions on current sip Use [email protected] for new developments on the application of sip
Bad Call.pcap
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Good Call.pcap
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